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Summary:ASTERISK-22005: Allow a sip peer to accept both AVP and AVPF calls
Reporter:Torrey Searle (tsearle)Labels:
Date Opened:2013-07-03 11:00:03Date Closed:2013-10-25 11:11:01
Priority:MajorRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:11.4.0 Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) 0024_optional_avpf.patch
( 1) optional_avpf_trunk.patch
Description:This patch adapts the behaviour of avpf to only impact the format of outgoing calls.  For inbound calls, both AVP and AVPF calls will be accepted regardless of the value of avpf in the configuration.  (This behavior was suggested on the asterisk-dev mailing list)
Comments:By: Matt Jordan (mjordan) 2013-07-09 08:56:28.689-0500

Hey Torrey -

I'm good with the behavior change as discussed on the {{asterisk-dev}} list. Other than a few minor formatting quirks ({{if(req->method != SIP_RESPONSE) {}} should have a space between the {{if}} and the {{(}}), the patch looks good to me as well.

Arguably, this is a behavioral improvement and not a bug fix however - which usually means it would go into trunk. Is there a case to be made for including this in 1.8/11?

By: Torrey Searle (tsearle) 2013-07-09 09:09:16.241-0500

I currently depend on this to get WebRTC working in the case that an upstream proxy is accepting both WebSocket and and SIP traffic (so both RTC and standard sip arrives on the same peer in asterisk)

That being said I'm happy to keep the patch on my ast11 if you prefer to keep it in trunk only.

By: Torrey Searle (tsearle) 2013-07-09 09:26:09.699-0500

trunk version of patch (with formatting quirk fixed

By: Matt Jordan (mjordan) 2013-07-09 10:31:14.016-0500

Hm... I'd say for interoperability concerns, it might be worth getting it in for 1.8/11 then. It's at least worth a discussion.

Thanks for the quick update!